The present invention relates to an apparatus, a method and an electroacoustic system for reverberation time extension.
From an acoustic point of view, a room might not be optimum for different applications. Thus, a musical performance normally necessitates some reverberation to sound good. On the other hand, speakers may partly not be understood when the room is too reverberant. Thus, an adaptation of the reverberation time by means of reverberation systems is useful.
For example, in theaters, conference centers, planetariums, seminar rooms, multifunctional rooms, different acoustic conditions may be needed for different situations, and in particular different requirements with regard to reverberation time are necessitated. For influencing the reverberation time, electroacoustic systems for reverberation time extension can be used. Such systems can either be incorporated into an already existing concert hall, however, it can also be useful to provide an electroacoustic system for reverberation time extension already when constructing and building respective buildings and halls, for example in exhibition construction. Reverberation time extension can also be desirable for audio reproduction for entertainment purposes.
In the following, the wave field synthesis technology will be discussed in more detail. The wave field synthesis (WFS) has been researched at TU Delft and presented for the first time in the late 80's (Berkhout, A. J.; de Vries, D.; Vogel, P.: Acoustic control by Wave-field Synthesis. JASA 93, 1993).
Due to the enormous requirements of this method with regard to computing power and transmission rates, wave field synthesis has so far been hardly applied in practice. Only the progress in the field of microprocessor technology and audio encoding allow today the usage of this technology in specific applications. First products in the professional area are expected next year. The basic idea of WFS is based on the application of the Huygen Principle of wave theory:
Each point captured by a wave is the starting point of an elementary wave spreading spherically or circularly. Applied to acoustics, any form of an incoming wavefront can be reproduced by a large number of loudspeakers arranged next to each other (in a so-called loudspeaker array). In the simplest case of an individual point source to be reproduced and a linear array of loudspeakers, the audio signals of each loudspeaker have to be fed with a time delay and amplitude scaling such that the radiated sound fields of the individual loudspeaker are superimposed properly. For several sound sources, the contribution to each loudspeaker is calculated separately for each source and the resulting signals are added. In a room having reflective walls, reflections can also be reproduced as additional sources via the loudspeaker array. Thus, the calculation effort depends heavily on the number of sound sources, the reflection characteristics of the recording room and the number of loudspeakers.
The advantage of this technology is in particular that a natural spatial sound impression is possible across a large area of the reproduction room. In contrast to the known technologies, direction and distance of sound sources are reproduced very exactly. To a limited extent, virtual sound sources can even be positioned between the real loudspeaker array and the listener.
Thus, by the technology of wave field synthesis (WFS), a good spatial sound can be obtained for a large range of listeners. As has been stated above, the wave field synthesis is based on the principle of Huygens, according to which wavefronts can be formed and built up by superimposing elementary waves. According to a mathematically exact theoretical description, an infinite amount of sources at an infinite small distance would have to be used for generating the elementary waves. Practically, however, a finite amount of loudspeakers are used at a finite small distance to each other. Each of these loudspeakers is controlled according to the WFS principle with an audio signal from a virtual source having a specific delay and a specific level. Level and delays are normally different for all loudspeakers.
As has been stated above, a wave field synthesis system operates based on the Huygen Principle and reconstructs a given waveform of, for example, a virtual source arranged at a specific distance to a listener by a plurality of individual waves. Thus, the wave field synthesis algorithm receives information on the actual position of an individual loudspeaker from the loudspeaker array to calculate then a component signal for this individual loudspeaker, which this loudspeaker then finally has to radiate, so that for the listener a superposition of the loudspeaker signal from the individual loudspeaker with the loudspeaker signals of the other active loudspeakers performs a reconstruction such that the listener has the impression that he is not exposed to sound from many individual loudspeakers but merely from a single loudspeaker at the position of the virtual source.
For several virtual sources in a wave field synthesis setting, the contribution of each virtual source for each loudspeaker, i.e. the component signal of the first virtual source for the first loudspeaker, the second virtual source for the first loudspeaker, etc. is calculated, to then add up the component signals to finally obtain the actual loudspeaker signal. In the case of, for example, three virtual sources, the superposition of the loudspeaker signals of all active loudspeaker would have the effect for the listener that the listener does not have the impression that he is exposed to sound from a large array of loudspeakers, but that the sound that he hears merely originates from three sound sources positioned at specific positions, which are equal to the virtual sources.
In practice, calculation of the component signals is performed mostly in that an audio signal assigned to a virtual source is provided at a specific time with a delay and a scaling factor, depending on the position of the virtual source and the position of the loudspeaker, in order to obtain a delayed and/or scaled audio signal of the virtual source, which represents the loudspeaker signal immediately when only one virtual source exists, or which, after addition with further component signals for the considered loudspeaker from other virtual sources, contributes to the loudspeaker signal for the considered loudspeaker.
Typical wave field synthesis algorithms operate independent of how many loudspeakers exist in the loudspeaker array. The theory underlying wave field synthesis is that each arbitrary sound field can be exactly reconstructed by an infinitely high number of individual loudspeakers, wherein individual loudspeakers are arranged infinitely close to one another. In practice, however, neither the infinitely high number nor the infinitely close arrangement can be realized. Instead, a limited number of loudspeakers exist, which are additionally arranged at specific predetermined intervals to one another. Thereby, in real systems only an approximation of the actual waveform is obtained, which would take place if the virtual source actually existed, i.e. were a real source.
Wave field synthesis means are further able to reproduce several different source types. A prominent source type is the point source where the level decreases proportionally 1/r, wherein r is the distance between a listener and the position of the virtual source. Another source type is a source radiating plane waves. Here, the level remains constant independent of the distance to the listener, since plane waves can be generated by point sources that are arranged at an infinite distance to each other.
After the above excursion on existing wave field synthesis means, we will now deal with systems for reverberation time extension known from conventional technology:
In US005109419A, Griesinger describes an electroacoustic system for reverberation time extension where different sound sources are recorded via microphone or direct input and are artificially reverberated via a reverb matrix. The output signals of this system are output to distributed loudspeakers and thus generate an artificial reverberation in the room.
Also, Poletti describes in “Reverberators for use in wide band assisted reverberation systems” US000000039189E an electroacoustic system for reverberation time extension based on the detection of spatial signals, and processing the same in a delay matrix which again controls a plurality of loudspeakers.
In US0051425869A, Berkhout describes an approach where a signal recorded in a room is convolved and reproduced via a reconstructed wave field.
In patent literature, there are different systems for reverberation time extension, such as U.S. Pat. No. 3,614,320 A and WO 2006092995 A1.
However, none of the systems allow flexible dynamic adaptation to different and alternating acoustic conditions and desires of the users concerning reverberation time extension.